Is it possible to set up good quality voice greetings in FreePBX without transcoding? After encoding .wav or .mp3 sound quality is greatly lost, especially with background music.
I dont have this problem, can you bypass the encoding? This has some good best practices.
Recording high quality voice prompts for Asterisk – Obviate.io
You are not going to improve on the codec used on the weakest link, which is often G.711 (A- or µ-law) for PSTN, and and a, possibly enhanced, GSM, for mobile phones.
In particular, if your callers are using mobile phones, don’t play music in the background. All mobile phone codecs are speech only and are likely to get really confused by background music.
Also note that .wav is a container format, not a codec algorithm. .wav files contain extremely low bit rate, speech only codecs, through full CD quality.
Asterisk .wav is 16 bit samples, uncompressed, and sampled at 8kHz. Asterisk. .WAV is GSM.
Unless you have a carrier operating on a fully GSM network this is no longer the case. There is now over 400 operators using IMS/VoLTE which uses either AMR-WB or AMR-NB. None IMS/VoLTE operators have moved to G722.
In those cases the issues that old GSM networks had for voice quality are gone.
AMR-WB is still a speech codec, not an audio codec. The Wikipedia entry qualifies the statement that it matches G.722 with “for clean speech”, which I assume means it will be compromised by background music.
It looks like you would need AMR-WB+ to get good speech over music, although the claim for music is qualified by “at low bit rates”, again presumably because it is a vocal tract model codec.
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